1. Field of the Invention
The present invention relates to a method, a system and the like concerning call control of a mobile communication network.
2. Description of the Prior Art
At present, there are widely available mobile communication services using portable terminals such as cellular phones or mobile phones. In the mobile communication, portable terminals can communicate each other via a relay device that is called a base station. Wireless communication is performed between the portable terminal and the base station. An area in which the portable terminal can communicate with the base station depends on the base station. If the portable terminal moves to the outside of the area, i.e., the outside of the communication area, the portable terminal has become a communication disable state. Although the mobile communication may be used as data communication, it is mainly used for audio communication (speech communication) by a telephone service in many cases. As a technique for improving convenience of such mobile communication, there is proposed a method as disclosed in Japanese unexamined patent publication No. 2002-320261, for example.
In the telephone service, a telephone set calls another telephone set on the other end, and the called side responds to the call so that speech communication can be realized between them. In the telephone service, this action making a state where speech communication is possible is referred to as “call establishment” or “audio path establishment” or the like. In other words, the call establishment enables start of speech communication. In addition, when the established call is disconnected (or finished), the speech communication is finished. Control concerning the call establishment and the disconnection is usually referred to as “call control” or “signaling” or the like.
Recently, a telephone service that is called an IP phone using a protocol of the Internet is used in many situations. The IP phone is a type of telephone services for speech communication via an IP network (IP phone network) that is made up of the Internet or an intranet or the like.
The IP phone uses a protocol that is called SIP (Session Initiation Protocol) for call control. The SIP is a protocol defined by RFC3261 or the like. According to the SIP, various types of message concerning call control are exchanged between portable terminals in order that call establishment or disconnection is made. Hereinafter, this message may be referred to as an “SIP message” in the description.
Here, a conventional process flow for establishing call by the SIP will be described with reference to FIG. 8. “SIP” in FIG. 8 indicates an “SIP server” that is a server performing call control by the SIP.
When a user dials the number of the other side of speech communication by using his or her portable terminal (referred to as a “portable terminal A”), the portable terminal A transmits an SIP message of “INVITE” to a portable terminal on the other end (referred to as a “portable terminal B”) (#901). This INVITE is an SIP message for requesting a call establishment.
The SIP server usually has a function as a proxy server, and it relays and transmits the SIP message to a destination, which is exchanged between devices such as portable terminals. The SIP server relays the INVITE transmitted in the step #901, by the function as the proxy server, so that the INVITE is transmitted to the portable terminal B on the other end (#903).
When it is relayed, the SIP server sends the SIP message of “100 Trying”, which indicates that the INVITE is received, to the portable terminal A (#902).
When the portable terminal B receives the INVITE, it transmits the “100 Trying” to the SIP server first (#904). Then, it transmits the SIP message of “180 Ringing”, which indicates that the user is being called, to the portable terminal A (#905). The SIP server relays the “180 Ringing” and transmits it to the portable terminal A (#906).
When a user of the portable terminal B responds to the call, the portable terminal B transmits the SIP message of “200 OK” to the portable terminal A (#907). The “200 OK” is the SIP message indicating that the procedure (or the process) is successful and means a response to the INVITE in this case. The SIP server relays the “200 OK” and transmits it to the portable terminal A (#908).
Note that in the SIP the portable terminal that issues the SIP message such as the INVITE for a request (the portable terminal A in this case) may be referred to as a UAC (User Agent Client). In addition, the portable terminal that responds to the SIP message (the portable terminal B in this case) may be referred to as a UAS (User Agent Server).
When the portable terminal A receives the “200 OK”, it transmits the SIP message of “ACK” for confirming that the call establishment is completed to the portable terminal B (#909). The SIP server relays the “ACK” and transmits it to the portable terminal B (#910). By the process described above, call establishment is performed between the portable terminals A and B. Thus, speech communication can be started.
When the speech communication is finished and one of the users does an operation for disconnecting the connection, a process for disconnecting the call (a disconnection process) is started. Here, a conventional process flow of disconnecting the call will be described with reference to FIG. 9. Note that it is supposed that the user of the portable terminal A does the operation for the disconnecting the connection.
When the user does the operation for the disconnecting the telephone, the portable terminal A transmits the SIP message of “BYE” to the portable terminal B (#922). The “BYE” is the SIP message for requesting disconnection (end) of the call.
The SIP server relays the “BYE” and transmits it to the portable terminal B (#924). In addition, in parallel with it or before or after it, resources that had been secured for the process for the portable terminal A in the call control are released (#923)
When the portable terminal B receives the “BYE”, it transmits the “200 OK” to the portable terminal A (#925). The SIP server relays the “200 OK” and transmits it to the portable terminal A (#927). In addition, in parallel with it or before or after it, resources that had been secured for the process for the portable terminal B in the call control are released (#926). When the portable terminal A receives the “200 OK”, the process of disconnecting the call (a release process) is completed.
In addition, the call control may be performed in cooperation between the SIP server and an application server that provides an application program concerning a service such as the IP phone, a videoconference or the like using the IP network. FIG. 10 shows a conventional process flow of call disconnection using the application server. Note that “AS” in FIG. 10 indicates the application server.
When the application server is used, the portable terminal A transmits the “BYE” to the application server via the SIP server (#932 and #933 in FIG. 10). The application server that has received the “BYE” releases resources that have been secured when the call establishment was performed (#934 and #938), and it transmits the “BYE” for informing that the call will be finished to the portable terminal B via the SIP server (#939)
In the mobile communication, if one of the portable terminals that are doing speech communication moves to the outside of the communication area or due to other factor, a communication disable state may occur. Then, the application server detects the communication disable state, and the call disconnection process is started. FIG. 11 shows the conventional process flow of the call disconnection when a communication disable state is detected. In this case, after the application server detects a communication disable state, the application server transmits the “BYE” to both the portable terminals (#954 and #960).
The IP phone performs the call establishment and the disconnection process in the procedure as shown in FIGS. 8-11, so that speech communication between the portable terminals is realized.
However, there is a case where there is no response to the SIP message that has been transmitted from the SIP server to the portable terminal due to a certain factor. In this case, the SIP server cannot do a process to be performed subsequently to the response. Therefore, in the conventional method, the SIP server retransmits the SIP message to the portable terminal for requesting a response again as shown in FIG. 12. If the retransmission is repeated for a predetermined time period and there is no response from the portable terminal, a process to be performed after the response is started.
However, in the conventional method described above, if a response (“200 OK”) is not returned concerning the “BYE” that was transmitted from the SIP server to the portable terminal, the next process cannot be started promptly. As a result, release of resources is delayed. In this case, use efficiency of the SIP server is lowered. In addition, if the retransmission of the SIP message to the portable terminal is repeated, it will cause an increase of traffic.